Compensate for the compression gain in the IntelligibilityEnhancer

webrtc/modules/audio_processing/intelligibility/intelligibility_enhancer_unittest.cc

 /*
  *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 219 matching lines @@
 
   intelligibility_enhancer->ProcessRenderAudio(
       render_audio_buffer->split_channels_f(kBand0To8kHz),
       IntelligibilityEnhancerSampleRate(sample_rate_hz),
       render_audio_buffer->num_channels());
 
   noise_suppressor->AnalyzeCaptureAudio(capture_audio_buffer);
   noise_suppressor->ProcessCaptureAudio(capture_audio_buffer);
 
   intelligibility_enhancer->SetCaptureNoiseEstimate(
-      noise_suppressor->NoiseEstimate());
+      noise_suppressor->NoiseEstimate(), 0);
 
   if (sample_rate_hz > AudioProcessing::kSampleRate16kHz) {
     render_audio_buffer->MergeFrequencyBands();
   }
 }
 
 // Processes a specified amount of frames, verifies the results and reports
 // any errors.
 void RunBitexactnessTest(int sample_rate_hz,
                          size_t num_channels,
@@ skipping 198 matching lines @@
 }
 
 TEST(IntelligibilityEnhancerBitExactnessTest, DISABLED_Stereo48kHz) {
   const float kOutputReference[] = {-0.009276f, -0.001601f, -0.008255f,
                                     -0.012975f, -0.015940f, -0.017820f};
 
   RunBitexactnessTest(AudioProcessing::kSampleRate48kHz, 2, kOutputReference);
 }
 
 }  // namespace webrtc